FreeSWITCH 是一个电话的软交换解决方案,包括一个软电话和软交换机用以提供语音和聊天的产品驱动。它可以用作交换机引擎、PBX、多媒体网关以及多媒体服务器等。可用于创建音、视频以及短消息类产品和应用,它采用 Mozilla Public License (MPL)授权协议。它的核心库libfreeswitch可以嵌入其它系统或产品中,也可以做一个单独的应用存在。
FreeSWITCH 支持多种通讯技术标准,包括 SIP, H.323, IAX2 以及 GoogleTalk ,可以方便的与其他开源的PBX系统进行对接,例如 sipX, OpenPBX, Bayonne, YATE 或者 Asterisk。它支持许多高级的 SIP 特性,例如 presence/BLF/SLA 、TCP TLS 和 sRTP,它还可以用来作为类似于 SBC (Session Border Controller) 的透明代理。

FreeSWITCH is a scalable open source cross-platform telephony platform designed to route and interconnect popular communication protocols using audio, video, text or any other form of media. It was created in 2006 to fill the void left by proprietary commercial solutions. FreeSWITCH also provides a stable telephony platform on which many telephony applications can be developed using a wide range of free tools.
FreeSWITCH was originally designed and implemented by Anthony Minessale with the help of Brian West and Michael Jerris. All 3 are former developers of the popular Asterisk open source PBX. The project was initiated to focus on several design goals including modularity, cross-platform support, scalability and stability. Today, many more developers and users contribute to the project on a daily basis.
We support various communication technologies such as Skype, SIP, H.323 and GoogleTalk making it easy to interface with other open source PBX systems such as sipXecs, Call Weaver, Bayonne, YATE or Asterisk.
FreeSWITCH supports many advanced SIP features such as presence/BLF/SLA as well as TCP TLS and sRTP. It also can be used as a transparent proxy with and without media in the path to act as a SBC (session border controller) and proxy T.38 and other end to end protocols.
FreeSWITCH supports both wide and narrow band codecs making it an ideal solution to bridge legacy devices to the future. The voice channels and the conference bridge module all can operate at 8, 12, 16, 24, 32 or 48 kilohertz and can bridge channels of different rates. The G.729 codec is also available under a commercial license.
FreeSWITCH builds natively and runs standalone on several operating systems including Windows, Max OS X, Linux, BSD and Solaris on both 32 and 64 bit platforms.
FreeSWITCH supports FAX, both over audio and T.38, and can gateway between the two.
Features
* Runs on Win32/MAC/UNIX
* IVR Application API
* 8kHz / 12kHz / 16kHz / 24kHz / 32kHz / 48kHz Audio
* Soft Conferencing
* Broadsoft SCA/SLA
* SIP B2UA/SRTP/TLS
* SIP BLF/SLA/PBX Features
* Presence
* GoogleTalk
* IPv4/IPv6
* ENUM/ISN
* Async Audio
* Event/Logger Engine
* Real Time
* zRTP (libzrtp)
* Many More
该版本修复了大量的 bug,推荐在生产环境中使用该版本。
最新版本:1.6.20
这是一个维护版本,修改包括以下几方面:
为模块 mod_av 增加了一条命令,以方便修改日志级别;
修复部分不匹配的编码解码器导致的错误响应;
修复使用 srtp/crypto 时导致的崩溃问题;
修复错误内存分配问题;
更详细的更新修改列表,请参见此处。
官方主页:http://www.freeswitch.org/